The present invention is related to the field of communications. In particular, the present invention is related to a method and apparatus for transcoding video and audio signals. More particularly, the invention provides a method and system for transcoding information (e.g., video, voice, data) from a first format to a destination format using a proxy transcoder server having a plurality of transcoding processes, where at least one is selected for transcoding the information. Merely by way of example, the invention is applied to a wide area telecommunication network, but it would be recognized that the invention can also be applied across many different types of multimedia protocols over transport networks such as the Internet, a mobile network, a local area network, PTSN, ISDN, SONET, DWDM, and others.
Telecommunication techniques have improved dramatically over the years. Many different types of networks such as fixed switched, packet based, wireless and mobile have been deployed. One of the most widely known world wide network called the “Internet” has popularized networking to many people around the world. An increase in use of wide area networks such as the Internet has resulted in many new on-line services such as electronic mail, video telephony, video streaming, electronic commerce, and others. Although computers originally connected to the Internet, other devices such as mobile phones, personal digital assistants, laptop computers, and the like have also been connected. Accordingly, many different types of devices now have access to many different types of services over a variety of networks.
A variety of network elements make up the networks, which connect the aforementioned devices together. Such devices are often connected by gateways and switches that handle transfer of data and conversion of messages from protocols of a sending network to protocols used by a receiving network. Gateways and switches convert analog voice messages to digital formats including G.711 and G.723.1, which are ITU standards. Gateways transmit the converted messages typically in a way similar to transmission of voice over IP. G.711 is an ITU standard for speech codecs that provides audio signals at 64 Kbps using either the A-Law PCM method or the mu-Law PCM method. G.723.1 is an ITU standard for speech codecs optimized for narrow-band networks, including Plain Old Telephone Systems and narrow band Internet connections. The standard uses the LD-CELP method and provides audio signals at 5.3 or 6.3 Kbps. Depending upon the application, there can be many others as well.
As merely an example in FIG. 1, a conventional system 100 is shown. This diagram is merely an example and provided for illustrative purposes only. A message originates from a mobile device 105, which is coupled to a wireless network. The message is sent from the mobile device to base station 110 through the wireless network. The base station is coupled to a service station 115, which is coupled to gateway 120. The base station receives the radio message from the mobile device 105, and converts the message, without transcoding, into a digital format, and transmit it to the service station 115. The reformatted message is subsequently transmitted to the gateway which in turns transmit the message to its destination, a user 140, through the Internet 125 and also through a variety of network elements. Such elements may include a gateway 130, a server 135, and others.
One or more gateways may also convert videoconferencing signals from one digital format to another, such as from H.320 to H.323, and transmit converted signals over the Internet. H.320 is an ITU standard for videoconferencing over digital lines, and it uses the H.261 video compression method, which allows H.320-compliant videoconferencing and desktop systems to communicate with each other over ISDN, switched digital lines and leased lines. H.323 is an ITU standard for real-time, interactive voice and videoconferencing over LANs and the Internet. Widely used for IP telephony, H.323 allows any combination of voice, video and data to be transported. H.323 specifies several video codecs, including H.261 and H.263, and audio codecs, including G.711 and G.723.1. Unfortunately, the audio and video standards have grown well beyond H.320, H.323, G.711, and G.723.1. That is, the proliferation of different standards has caused difficulty in communicating messages between them. Additionally, any communication between such standards has caused a proliferation of complex conversion techniques, which are time consuming and lack efficiency. Accordingly, there is a need for an efficient way to convert information or transcode between various formats in real time. Because some systems such H.320 and H.324 are circuit switched systems (data is transmitted as a continuous stream of bits) and some other systems are packet based, the connection of circuit-based to paket based systems require the demultiplexing of bits from circuit based bitstreams into packet (circuit-to-packet) and vise versa (packet-to-circuit). Note that different system protocols such as H.320, H.323, H.324, 3GPP-324M, SIP and SDP, make use of different signaling methods (to setup connections and exchange terminal capabilities). The inter-connectivity of these systems require the trans-signaling and the converstion of terminal capabilities so terminal can understand what terminals using different protocols are capable of.
From the above, it is seen that an improved way of transferring information from a source to a destination is highly desirable.